The present invention is directed to a communications system for communicating voice, video or data over a distributed computer network, such as the global Internet. More particularly described, the present invention provides a system for predicting, in real time, the quality of multimedia communications over the Internet.
The Internet can support the communication of multimedia data, including voice, video and data communications, via a global network of distributed computers without relying on the Public Switched Telephone Network (PSTN). FIG. 1 is a block diagram showing one example of an Internet-based telephony service, which can be supported by a communications system 100. This implementation of the communications system 100 is described in more detail in U.S. patent application Ser. Nos. 09/154,564 (now U.S. Pat. No. 6,426,955) and 09/154,566 (now abandoned), which are assigned to the assignee for the present application and fully incorporated herein by reference. A user 112 in the United States places a telephone call to a party 124 in France. Instead of using the international long distance network, however, the user 112 makes a local call via a local telephone network 114 in the United States to an Internet telephony endpoint or gateway 116. The gateway 116 converts the user""s voice-band communications to digital packets and transfers those packets across the Internet 118 to a peer gateway 120 located in France. The gateway 120 performs the reverse actions of the gateway 116, converting the digital packets back to voice-band communications and delivering them to the called party 124 through the local telephone network 112 in France.
Before such Internet-based telephony services can achieve widespread commercial success, however, they must be able to assure end users of an acceptable level of quality for their conversations. The public Internet at present relies on a packet-based technology to support the communication of information via a multi-point computer network. As such, this computer network cannot offer the same certainty of quality as the legacy telephone network. In general, Internet-based communications may encounter latency, variable delay, and packet loss. These factors, along with other network conditions, may detract from the quality of multimedia communications for an Internet-based telephony service. The Internet telephony industry has recognized that service quality is a major impediment to commercial success. Consequently, industry members have either proposed or adopted a variety of strategies to address the problem of providing an acceptable level of quality for Internet-based telephony communications.
For example, a distributor for Internet telephony services can offer its customers and suppliers the option of using a dedicated communication facility that connects directly to a private communication network. This approach bypasses the publicly available resources of the Internet, while providing a more controlled communications environment that can reduce the uncertainties associated with Internet-based communications. This private network approach is economically inefficient, however, because it requires dedicated links to the private network rather than a link to the nearest connection point for the public Internet. At present, these private network links only carry Internet telephony traffic while, in contrast, a general Internet connection can support access to the World Wide Web, electronic mail, file transfer, and other conventional distributed computer network operations.
Another approach to the provision of service quality by the Internet telephony industry is the use of a nationwide fiber backbone to provide Internet telephony services. A major communications carrier can tightly control the use of its nationwide fiber backbone, thereby ensuring a certain level of quality of communications carried by that network. This approach, however, is limited by the extent of the backbone network, which typically provides connections only to selected urban regions. In comparison to the existing global distributed computer network of the public Internet, the expansion of a single fiber backbone network to provide ubiquitous world-wide communications service is an impractical solution.
The Internet Engineering Taskforce has proposed a special communications protocol, the Resource Reservation Protocol (RSVP), which provides for the reservation of resources on the Internet. It will be appreciated that multimedia communications can be assured of a certain minimum quality if sufficient Internet sources are reserved in advance of those communications. To be effective, however, RSVP must be deployed in all devices in the path of such multimedia communications. Consequently, this approach requires a comprehensive upgrade of all current devices in the Internet, which is a less than a practical near-term solution for a desired quality level of Internet multimedia communications.
Another service quality approach, proposed by Cisco Systems, Inc., is the use of a proprietary technology based on weighted fair queuing to ensure the quality of Internet-based communications. For effective operations, components utilizing the weighted fair queuing technology must be deployed throughout the entire communications path. This approach requires provisioning the entire public Internet with equipment utilizing the weighted fair queuing technology, which is at present offered only by Cisco Systems. Consequently, the marketplace has not accepted this approach as a wide scale solution to the service quality issue for Internet telephony communications.
In view of the foregoing, there is a need to characterize communications across the public Internet rather than private networks. Moreover, there is a need to support a level of quality communications across multiple administered domains rather than a single domain. There is a further need for an Internet-based communications system that operates with existing Internet infrastructure without requiring a comprehensive upgrade of existing equipment. In summary, there is no present mechanism to quantitatively assess the impact of latency, variable delay, and packet loss factors on communications quality, nor is there a commercially feasible method of predicting those factors in advance of a particular communication. The present invention provides a novel and non-obvious technical solution that addresses both of these needs of the prior art.
The present invention provides a system for predicting the quality of a communication carried via a distributed computer network, such as the global Internet or a frame network, before the initiation of that communication between a pair or endpoints for a communication system. The inventive prediction system comprises software-implemented processes that can be installed within a gateway of an Internet telephony system or in an external system that provides prediction results to the Internet gateway. A first process accepts historical data and current environment data and, in response, generates a prediction of objective performance characteristics. A second process accepts the prediction of objective performance characteristics and, in response, generates an estimate of an expected quality of communications with a called party. This user quality estimate is typically used to support a determination of whether to proceed with the communication or to rely upon an alternative to a communication completed by an Internet telephony system.
The first process of the prediction system, which collects historical data components for prior communications, can rely upon different communication protocols to collect Internet communication measurements and environment attributes. The International Telecommunications Union (ITU) H.323-series protocols for multimedia communication can be used to complete a direct measurement of a round-trip time between communicating endpoints. The Internet Control Measurement Protocol (ICMP) supports the use of ICMP path probes during a multimedia communication to collect fine-grained network path information for that communication. The Border Gateway Protocol (BGP) can support the collection and recording of BGP peering information during a multimedia communication to discover coarse-grained network path information for that communication. For example, BGP peering information can be collected from the nearest transit autonomous system to support the extrapolation of coarse-grained path information for stub autonomous systems. Consequently, the first software-implemented process of the prediction system can use both fine-grained path information and coarse-grained path information as indicators of the quality of service for multimedia communications.
While historical data provides information about prior communications, current environment data defines information about a specific communication under consideration. The information for a potential communication typically includes the identities of the endpoints, the current time and date, the coarse-grained paths between the endpoints, and a single sample of round-trip delay from the initiating endpoint. The fine and coarse-grained path measurement systems described above can be used to support the collection of current environment data that represents an input to the first process of the prediction system.
The first software-implemented process of the prediction system, the objective performance prediction process, can be constructed by the combination of a genetic algorithm and a neural network. The genetic algorithm is typically used to select significant historical data, whereas the neural network supports the prediction of current performance for a communication. The genetic algorithm accepts historical data and current environment data and outputs relevant historical data. The genetic algorithm can be tuned to select historical data based on selection criteria comprising similarity of network paths, identity of endpoints and time/date information. The neural network accepts current environment data and relevant historical data and, in response, generates a prediction of objective performance for the communication. In particular, the neural network can be implemented by an optimal interpolative neural network that supports the prediction of current performance for a current or future communication based on significant historical data from previous communications.
The second part of the software-implemented process of the prediction system provides an estimate of the subjective user quality based on objective performance prediction. This estimation process can be implemented according to the calculations and algorithms described in ITU-T Recommendation G. 107 (December 1998), xe2x80x9cThe E-Model, a Computational Model for use in Transmission Planning,xe2x80x9d and ETSI Guide EG 201 377-1 V1.1.1 (1999-01), xe2x80x9cSpeech Processing, Transmission and Quality Aspects (STQ); Specification and Measurement of Speech Transmission Quality; Part 1: Introduction to Objective Comparison Measurement Methods for One-Way Speech Quality Across Networks.xe2x80x9d Alternatively, the estimation process can be implemented by a conventional neural network that determines a subjective quality of a communication, as perceived by a human user, based on objective measurements or predictions of fraction packet loss or round-trip delay. This neural network can include (1) inputs defined by the fraction of packets lost in each path direction and characterizations of round-trip delay and (2) outputs representing an estimate of subjective user quality.